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After ~2 months of testing, stress testing and tuning, OpenSIPS 1.8 moves from beta to full stable stage!

Hello,

We are proud to announce that the new release of XiVO Skaro is available on our mirror.
This is mostly a bug fix release, with 40 bugs closed.

New features¶

• Screen popup management (or "sheets") has been cleaned and documented
• A counter for logged agents per queue has been added to the XiVO Client
• SCCP multi-line support
• Packaged Asterisk 1.8.11.1
• Ability to logoff agent automatically
• LDAP lookup for phone numbers

As usual, you must upgrade your XiVO by following this procedure: http://documentation.xivo.fr/stable/upgrade/upgrade.html

List of fixes and features: https://projects.xivo.fr/versions/149
XiVO tests report (french version) http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/tests-report-xivo-1.2.7.pdf

XiVO Client¶

The address to get the XiVO Client has changed : you can download the XiVO client from http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/
We have set up a redirection from the old location.

Direct links :

• Debian / Ubuntu 32 bits : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/xivoclient-1.2.7.3-i386.deb
• Debian / Ubuntu 64 bits : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/xivoclient-1.2.7.3-amd64.deb
• Windows : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/xivoclient-1.2.7.3-x86.exe
• Mac OS : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/xivoclient-1.2.7.3.dmg

XiVO client tests report from http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/tests-report-xivo-client-1.2.7.pdf

Download it¶

You can download this version from : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.7/squeeze-xivo-skaro-1.2.7.iso
or use our installation script : http://mirror.xivo.fr/fai/xivo/xivo_install_skaro.sh (see http://documentation.xivo.fr/stable/installation/installation.html)

Thank you.

The XiVO dev team

Today, if you have to manage Video Communications over Internet (public, private, wifi, 3g, 4g…), you may think about how to solve some very common public IP networks’ barriers and issues like: Packet lost, Firewalls policies, Router NAT, Packets filtering, Low Bandwidth, Connection cuts off…

Now, the question is:
“How we really manage to make voice or video calls on that network conditions!”

Obviously, it’s not so easy with a common telecom protocol.
This SIP vs RTMP Comparison Report, will help you to learn a bit more about these two protocols:

1. The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP)…
2. The Real Time Messaging Protocol (RTMP) is a public protocol developed by Adobe/Macromedia for streaming audio, video and data over the Internet, and not only for a Flash player…

You can get our complete comparision SIP vs RTMP (for videocalls) report here.

Download: report-i6net-sip-rtmp-04-2012.pdf - 24 hits

Authors: I6NET Solutions and Technologies, SL
Date: April, 27 2012

 

We are pleased to unveil the first IVR / IVVR demo service to test 10 TTS engines!

This demo show you how each TTS works with our VXI* VoiceXML browser to test many languages by yourself over a PSTN call or a web call. Our VXI* server is connected to all of these engines at the same time, and it’s running each VoiceXML session in real-time. We will extend this demo service for more languages and voices in the next weeks.

We want to thank all our TTS technology partners for their engine contribution on this demo service where you can select and discover all their product’s voices.

Text-to-Speech Engines:

• Verbio
• Ivona
• Google
• Voxygen
• Loquendo-Nuance
• Truevoice-SpeechTechnologyGroup
• Lumenvox
• Acapela-Group
• MBROLA-eSpeak (coming soon)
• Festival Lite (coming soon)

Languages:

• English (EN)
• English UK (EN-UK)
• English US (EN-US)
• French (FR)
• Spanish (ES)
• Polish (PL)
• German (DE)
• Arabic (AR)

PSTN Phone Number:
Call +34 911877474 (local geographic) from any phone or mobile

Web Phone (Audio):
http://demo.i6net.org/phone/audio.html   Call 0013  (register with your own nickname)

Web Phone (Video):
http://demo.i6net.org/phone  Call 0013  (register with your own nickname)

Access Codes (4 digits PIN):

Verbio TTS
8101 (EN), 8102 (ES)

Ivona TTS
8201 (EN), 8202 (FR), 8103 (ES), 8204 (PL), 8205 (DE)

Google TTS
8301 (EN), 8302 (FR), 8303 (ES)

Voxygen TTS
8401 (EN), 8402 (FR), 8403 (ES)

Truevoice TTS
8501 (EN-UK), 8502 (EN-US), 8503 (ES)

Loquendo-Nuance TTS
8601 (EN), 8602 (FR), 8603 (ES)

Lumenvox TTS
8701 (EN-UK), 8702 (EN-US), 8703 (ES)

Acapela TTS
8801 (EN), 8802 (FR), 8803 (ES), 8804 (DE), 8805 (AR)

Coming soon:

MBROLA-eSpeak TTS
9001 (EN), 9002 (FR), 9003 (ES)

Festival-Lite TTS
9101 (EN)

 

Hello,

We are proud to announce that the new release of XiVO skaro is available on our mirror.

This is mostly a bug fix release, with 42 bugs fixed.

New features in this release:

• agent logout and agent pause action on CTI presence update
• attended transfer via the contact xlet

As usual, you must upgrade your XiVO by following this procedure: http://documentation.xivo.fr/stable/upgrade/upgrade.html

List of fixes and features: https://projects.xivo.fr/versions/148
XiVO tests report (french version) http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.6/tests-report-xivo-1.2.6.pdf

You can download the XiVO client from http://downloads.xivo.fr/xivo_cti_client
XiVO client tests report from http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.6/tests-report-xivo-client-1.2.6.pdf

You can download this version from : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.6/squeeze-xivo-skaro-1.2.6.iso

or use our installation script : http://mirror.xivo.fr/fai/xivo/xivo_install_skaro.sh (http://documentation.xivo.fr/stable/installation/installation.html)

Thank you.

The XiVO dev team

I'm glad to announce the OpenSIPS development Course - a training course which is addressed to people who want to develop (C development) new modules or functionalities for OpenSIPS 1.x.

This is a free online course, spread across several sessions with different topics.

Newfies-Dialer is a free and open source multi-tenant voice broadcasting solution and auto dialer system to enable the automated delivery of interactive phone calls to contacts, clients and the general public, making it perfect for telemarketing, mass emergency broadcasting, phone polling, surveys and voting.

There are a number of enhancements to this latest version of Newfies-Dialer, including scalability, speeding up the import and campaign start process, allowing duplicate contacts on a campaign, adding Chinese and Turkish to the interface in addition to a round of bug fixes.

More information about Newfies-Dialer can be found at http://www.newfies-dialer.org/ and there is a beginners guide to installation and operation at http://www.newfies-dialer.org/documentation/beginners-guide/.

Alternatively, professional installation, training and support is available. For more details, see http://www.star2billing.com/products/auto-dialer-platform/

YoursThe Star2Billing team

Hello,

We are proud to announce that the new release of XiVO skaro is available on our mirror.

Berofos integration, 37 bugs fixed.

You must upgrade your XiVO by following this procedure: http://documentation.xivo.fr/stable/upgrade/upgrade.html

List of fixes and features: https://projects.xivo.fr/versions/147
XiVO tests report (french version) http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.5/tests-report-xivo-1.2.5.pdf

You can download the XiVO client from http://downloads.xivo.fr/xivo_cti_client
XiVO client tests report from http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.5/tests-report-xivo-client-1.2.5.pdf

You can download this version from : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.5/squeeze-xivo-skaro-1.2.5.iso

or use our installation script : http://mirror.xivo.fr/fai/xivo/xivo_install_skaro.sh (http://documentation.xivo.fr/stable/installation/installation.html)

Thank you.

Nico & the dev team

Hello,

We are proud to announce that the new release of XiVO skaro is available on our mirror.

SCCP integration in web interface, 30 bugs fixed.

Be aware that our repositories GPG key has changed, so you must upgrade your XiVO by following this procedure: http://documentation.xivo.fr/stable/upgrade/upgrade.html

List of fixes and features: https://projects.xivo.fr/versions/146
XiVO tests report (french version) http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.4/tests-report-xivo-1.2.4.pdf

You can download the XiVO client from http://downloads.xivo.fr/xivo_cti_client
XiVO client tests report from http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.4/tests-report-xivo-client-1.2.4.pdf

You can download this version from : http://mirror.xivo.fr/iso/squeeze-xivo-skaro-1.2.4/squeeze-xivo-skaro-1.2.4.iso

or use our installation script : http://mirror.xivo.fr/fai/xivo/xivo_install_skaro.sh (http://documentation.xivo.fr/stable/installation/installation.html)

Thank you.

Nico & the dev team

We are very pleased to present the 7.0 release ref. 2012-03-22 of our VXI* VoiceXML Browser for Asterisk. This new 2012 edition is the most advanced interpreter that can run over Asterisk . As in previous releases, we include specific improvements and new features that will provide you the best IVR / IVVR experience for your voice and video telephony projects. VXI* 7.0 is suitable for production platforms running with all latest Asterisk 1.4, 1.6, 1.8 kernels! (all Asterisk’s packages are available for download too).

Many Linux OS Distributions versions available based on:

• Debian 6.0 Squeeze 32bit | 64bit
• Debian 5.0 Lenny 32bit | 64bit
• CentOs 6.0-6.2 32bit | 64bit
• CentOs 5.4-5.7 32bit | 64bit

You can download these new binary packages from this website for registered users.

Powered by I6NET Software

New features added and modifications:

• add: Prefix ‘@’ in the account number to select by caller number.
• mod: Correction for S_ISREG symbol not found (load module).
• add: Special mode for <clear>, namelist with “.” not clear eventcounters.
• add: Account parameter ‘wait’, wait before start the VoiceXML browser.
• add: Option account force ‘vxml’ to disable ‘@’ execution.
• add: Prefix “record:” to access to the recorded local messages (record dest).
• add: Add attribute ‘dest’/'destexpr’ to the record (record in recorddirectory).
• add: Parameter to control the originate delay.

Thank you for your continuous support!

One more half an year, one more major release - OpenSIPS 1.8.0 beta release is out

OpenSIPS 1.8.0 comes with several major improvements (dialog support, TCP, security and validation), but also with new functionalities (like CacheDB Interface - REDIS/CASSANDRA, DNS caching, SIPcapturing, RabbitMQ support, LUA support, etc).

Hello,

There is a new gpg key for our mirrors. To install the key on your system, execute this line on your cli:

wget http://mirror.xivo.fr/xivo_current.key -O - | apt-key add -

Thanks.

Nico & the dev team

We are pleased to reference a new research paper from the Department of Computer Science, University of Oviedo, Spain.

This paper provides an insight into the process of developing comprehensive voice-enabled and interoperable learning services. It provides a background on the Voice Interactive Classroom, a service-oriented architecture that enables cross-platform multichannel access to Internet-based learning. Strategies and practices for designing, developing, documenting and evaluating service-oriented applications are also presented, which allow aural access to existing e-learning platforms. In addition, it’s summarise the development of four voice-enabled learning services and report the findings obtained from evaluating their usability and didactic effectiveness. This research is intended to facilitate students and practitioners with planning, implementing and evaluating their own solutions.

This research paper proposes the use of a service-oriented approach, supported by the e-learning specifications IMS Abstract Framework and The Open Knowledge Initiative, to provide a complementary audio communication channel and enable the interoperability of audio features among the various e-learning platforms and components. In addition, the use of the W3C recommendations of VoiceXML, along with the methods and techniques of adaptive educational hypermedia, allows overcoming the limitations of using unusual or non-standard languages to describe voice dialogues and better adapt the on-line learning process to innovative e-learning scenarios.

“We introduced the use of I6NET VXI* VoiceXML browser, which proved easy to install and reliable in use. VXI* can be used altogether with Asterisk and various Text to Speech and Automatic Speech Recognition solutions, including a few which provide support for the use of the Spanish language. This characteristic makes it highly convenient for our developments and an important factor to choose it in following studies” said Víctor Manuel Álvarez García, Faculty Member Assistant Professor – European Ph.D. Learning Technologies.

Source: http://onlinelibrary.wiley.com/doi/10.1002/cpe.2814/abstract

Authors: Víctor Manuel Álvarez García, M. Puerto Paule Ruiz, Moisés Riestra González, Juan Ramón Pérez Pérez
Publication: Concurrency and Computation
Publisher: John Wiley and Sons, copyright 2012
Date: Feb 1, 2012